1. Field of the Invention
The present invention relates generally to the field of teleconferencing systems, and more particularly to communication of data during an audio conference.
2. Background of the Invention
The telecommunications industry is constantly creating alternatives to travel for reaching a meeting forum. Teleconferencing has enabled many users to avoid long and expensive trips merely to meet with others to discuss business related topics and make important decisions. In addition, teleconferencing often replaces face to face meetings involving even the shortest of trips, such as those involving office locations relatively close in distance.
While teleconferencing typically offers good sound quality, attempting to send data over the same audio channel frequently disrupts a teleconference by diminishing the sound quality. Data, such as call control signals, can be sent via in-band signaling. However, conventional in-band signaling typically interrupts a call.
In order for participants to receive any type of data related to the call or otherwise (e.g., exchanging information via electronic mail (E-mail) while on the call), a separate communication channel must be utilized according to some prior art systems. Disadvantageously, having more than one channel can be distracting to the participants. A further disadvantage is that some type of connection to a network must be established for each communication channel. Unfortunately, all rooms are not equipped with numerous network connection outlets. Further, participants may connect at different speeds, so that all participants may not receive data at the same time. Yet another disadvantage of requiring numerous network connections to gain access to data related to the teleconference is that participants may be tempted to perform other activities while connected to the network.
Further, businesses and organizations mainly have PSTN lines. However, not all businesses and organizations have other dedicated communication lines such as ISDN or LAN. Accordingly, transfer of data related to the teleconference generally may be impossible without call interruption in prior art systems.
While a variety of processes exist for sending data over an existing audio connection, these processes typically are unable to send the data without some disruption, require a great deal of signal processing, are expensive to implement, etc. For example, on-off keying of a carrier has long been used to carry digital data over an analog channel (e.g., Morse code). However, while this process requires minimal processing, it is extremely disruptive to an audio signal present in the same channel. Frequency-shift keying of a carrier has also long been utilized, but this process makes any other use of the audio channel virtually impossible. Using DTMF tones is a common technique on telephone lines, but using DTMF tones is also very disruptive to a conversation occurring via the telephone lines.
Various types of modems are capable of carrying data over an audio channel at rates of 200 bps to 56,000 bps, but this process fully occupies the audio channel, precluding other uses of the audio channel. Hybrid modems implementing “voice over data” have been implemented to share an audio channel with data, but these modems require intensive computational power to recover the data. Further, without processing, these hybrid modems render the audio channel unusable since the voice is digitized and sent as data, requiring the modem to operate, using a great deal of computational power. An unprocessed audio channel emits a loud noise, like the normal sound emitted by a data modem.
Therefore, there is a need for a system and method for providing data to teleconference participants without interrupting the call or requiring intensive processing of digital or analog signals. Furthermore, there is a need for a process for embedding a low data-rate data connection within an existing narrow-band connection with minimal processing.